Fixing leaked reference from SCTP transport to DTLS/ICE transport.
This was causing ICE pings to continue going out on PeerConnections
that use DataChannels, even after closing the PeerConnection.
This CL adds a two-line fix, and an integration test that will catch
this and similar issues.
Bug: webrtc:7655
Change-Id: I589a2a1aaf6433c1d65be69a1267e1b52a33534b
Reviewed-on: https://webrtc-review.googlesource.com/37145
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21488}diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index ee1cffe..1d96a7f 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -5102,6 +5102,8 @@
void PeerConnection::DestroySctpTransport_n() {
RTC_DCHECK(network_thread()->IsCurrent());
sctp_transport_.reset(nullptr);
+ transport_controller_->DestroyDtlsTransport_n(
+ *sctp_transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
sctp_content_name_.reset();
sctp_transport_name_.reset();
sctp_invoker_.reset(nullptr);
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index abad10f..0bc68a2 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -3659,6 +3659,31 @@
kMaxWaitForFramesMs);
}
+// Test that after closing PeerConnections, they stop sending any packets (ICE,
+// DTLS, RTP...).
+TEST_F(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
+ // Set up audio/video/data, wait for some frames to be received.
+ ASSERT_TRUE(CreatePeerConnectionWrappers());
+ ConnectFakeSignaling();
+ caller()->AddAudioVideoMediaStream();
+#ifdef HAVE_SCTP
+ caller()->CreateDataChannel();
+#endif
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+ ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount,
+ kDefaultExpectedAudioFrameCount,
+ kMaxWaitForFramesMs);
+ // Close PeerConnections.
+ caller()->pc()->Close();
+ callee()->pc()->Close();
+ // Pump messages for a second, and ensure no new packets end up sent.
+ uint32_t sent_packets_a = virtual_socket_server()->sent_packets();
+ WAIT(false, 1000);
+ uint32_t sent_packets_b = virtual_socket_server()->sent_packets();
+ EXPECT_EQ(sent_packets_a, sent_packets_b);
+}
+
} // namespace
#endif // if !defined(THREAD_SANITIZER)
diff --git a/rtc_base/virtualsocketserver.cc b/rtc_base/virtualsocketserver.cc
index d461bf1..d8771e7 100644
--- a/rtc_base/virtualsocketserver.cc
+++ b/rtc_base/virtualsocketserver.cc
@@ -834,6 +834,7 @@
int VirtualSocketServer::SendUdp(VirtualSocket* socket,
const char* data, size_t data_size,
const SocketAddress& remote_addr) {
+ ++sent_packets_;
if (sending_blocked_) {
CritScope cs(&socket->crit_);
socket->ready_to_send_ = false;
@@ -897,6 +898,7 @@
}
void VirtualSocketServer::SendTcp(VirtualSocket* socket) {
+ ++sent_packets_;
if (sending_blocked_) {
// Eventually the socket's buffer will fill and VirtualSocket::SendTcp will
// set EWOULDBLOCK.
diff --git a/rtc_base/virtualsocketserver.h b/rtc_base/virtualsocketserver.h
index e25a5ff..e17caae 100644
--- a/rtc_base/virtualsocketserver.h
+++ b/rtc_base/virtualsocketserver.h
@@ -148,6 +148,10 @@
bool CloseTcpConnections(const SocketAddress& addr_local,
const SocketAddress& addr_remote);
+ // Number of packets that clients have attempted to send through this virtual
+ // socket server. Intended to be used for test assertions.
+ uint32_t sent_packets() const { return sent_packets_; }
+
// For testing purpose only. Fired when a client socket is created.
sigslot::signal1<VirtualSocket*> SignalSocketCreated;
@@ -282,6 +286,9 @@
uint32_t delay_stddev_;
uint32_t delay_samples_;
+ // Used for testing.
+ uint32_t sent_packets_ = 0;
+
std::map<rtc::IPAddress, int> delay_by_ip_;
std::map<rtc::IPAddress, rtc::IPAddress> alternative_address_mapping_;
std::unique_ptr<Function> delay_dist_;
